Internet telephony or voice communication over IP networks (hereinafter, referred to as VoIP) may not be able to ensure Quality of Service (QoS) due to the possibility of excessive end-to-end delay, packet loss, high delay, jitter, and the like, because the Internet Protocol (IP) network is a best-effort network. Among these factors, voice delay is one of the most significant factors influencing the service quality. The user starts feeling uncomfortable with the delay longer than 200 ms and, if the delay becomes longer than 400 ms, most people complain of significant inconvenience. Accordingly, it is desirable to improve real-time characteristics of voice communication by reducing the end-to-end delay. The end-to-end delay roughly includes processing delay and network delay.
The processing delay is the total delay incurred by both the sender and receiver while processing the voice signals with the exception of the time the packet propagates the network. The processing delay includes the transmitter delay of encoding the voice signal input, through a microphone and transmitting encoded Real-Time Protocol (RTP) packet, and the receiver delay of decoding the received packet and outputting the decoded voice signal through a speaker. Research is being conducted for technologies to reduce the delays in capturing, encoding, decoding, and rendering the voice signal.
Recently, the VoIP service for the Android® operating system-based smartphone users is introduced by Social Network Service (SNS) providers as well as the network operators, and the service competition is becoming intense. In spite of this situation, the end-to-end delay of the conventional Android® operating system-based VoIP service does not meet the requirement of 400 ms regardless of the application and type of terminal, and this is the big obstacle for securing the VoIP service quality on the Android® platform.